similar to: Too Many Open Files, Hung SIP Sessions, Can I Increase File Count?

List users, Using Asterisk in an inbound call center environment has led us to pushing the limits of vertical scaling. In order to treat each caller fairly and to utilize our agents as efficiently as possible, it is desirable to configure each client as a single queue. As far as I know, Asterisk's queues cannot be distributed across servers, so the size of the largest queue we service

Another trivial question: Is there a "place" where all the parameters are documented ? In example (my example!) I would like to know the meaning of a lot of parameter that can be used in sip.conf, A lot of these keywords are intuitive keywords (i.e. NAT=YES/NO;PORT=5060; context=xxxxx) but other are not (at least for me) i.e.: type = peer, friend insecure=very host=dynamic and so on.

Hi all, I am trying to stream for over 1k users on Ubuntu 16.04. I notice that when stream connection is over 1024, it get warning like this: WARN connection/_accept_connection accept() failed with error 24: Too many open files Tried these configs and reboot, it won't work! /etc/pam.d/common-session session required pam_limits.so /etc/sysctl.conf fs.file-max = 100000

Hi All, I've been growing a large headache on this one, i have a number of LDAP servers behind loadbalancing, since 2 days i constantly get the error: Too many open files. Although I'm not a newbie with linux I'm unable to resolve this, I have took the following stept: Changed the /proc/sys/fs/file-max to 65535 Added the following configuration to /etc/security/limits.conf: ldap

Kapejod is working on a fix for the CDR problem in bristuff. See below ----- Forwarded message from kpj@junghanns.net ----- Resent-From: tzafrir.cohen@xorcom.com Resent-Date: Fri, 10 Feb 2006 13:01:25 +0200 Resent-Message-ID: <20060210110125.GU16880@xorcom.com> Resent-To: tzafrir@cohens.org.il Envelope-to: tzafrir.cohen@xorcom.com Delivery-date: Fri, 10 Feb 2006 05:19:50 -0500 Date: Fri,

Hello, As an exercice, I installed sipp on the same box as a Asterisk 11.4 instance (to keep network equipements out of the equation). I'm focusing on the maximum number of new calls this Asterisk instance can deal with. Here is the dialplan (AEL) I'm playing with: _X. => { Verbose(0,Incoming call from ${CALLERID(num)} to ${EXTEN} in ${CONTEXT} - case A);

I am getting an error about to many open files. I tried to "echo 500000 > /proc/sys/fs/file-max" The number is there now, but I continue to get the error. Is there something else to do? Jerry

Hi, Besides file-max and file-nr is there anywhere else I should be looking to solve a C program giving me 'too many open files' problem? (centos 3.4) While the program is complaining here were the values file-max 209632 file-nr 3655 258 209632 lsof | wc -l around 7000 during and about 1111 less after I closed the application. Any ideas? thx

Hi all, I just arrived in Italy from Cebit, qhere I spoke with digium and Beronet people. They told me to try to use the mISDN stack to drive beronet and the new upcoming digium ISDN Cards. SO I searched, find http://www.beronet.com/download/card_installation_guide.pdf, and I immediately got the error: asterisk01:~ # cd /usr/src/install-misdn/ asterisk01:/usr/src/install-misdn # make install

Is there any way to customize VoiceMail ? I would like to customize the message played to callers sent to the voicemail becouse the extension is busy or otherwise unavailable. Is it a way to record a welcome message and use it ? thanks in advance, Andrea Chi ricevesse questa mail per errore e' gentilmente pregato di cancellarla. Visitate il sito http://www.frameweb.it

I realize I cannot use inband audio for phones (voicemail and internal ivr, password for external trunks and other thing not working) So I put everywhere rfc2833. Doing this, anyway, make any EXTERNAL IVR NOT working. I see a lot of posts about this, but no solution, becouse using inband audio (which works for outside...) breaks inside IVR Is it possible to define to use inband audio ONLY on

Running across some curious stuff with ulimit on CentOS 5.10. We have a non CentOS packaged version of Asterisk (using their packages) that we start at boot time with a typical RC script. Recently it started whining that it couldn't open enough file handles. As we dug further into this, it appears that at boot time, it inherits ulimit from init, which is pretty low: 1024. We've set

Hello, we are curently benchmarking an asterisk system 1034 sip users are logged into this system and the test software is trying to establish 400 concurrent calls. In the CLI I see the following messages: Apr 19 14:20:51 WARNING[4045]: rtp.c:911 ast_rtcp_new: Unable to allocate socket: Too many open files Apr 19 14:20:51 WARNING[4045]: acl.c:306 ast_ouraddrfor: Cannot create socket Apr 19

I have a problem with CDR recording in Asterisk 1.2.x. This is the situation: An Asterisk 1.2.0 (BRIstuffed-0.3.0-PRE-1 with florz) machine with a single HFC-S ISDN BRI card. I log the call records to both the Master.csv and MySQL. The problem is that when an incoming call from the ISDN line is logged to the CDR, the "src" and the "clid" field show up as something like

Hi to all I've mada up my mind and i tried to change from i4l to chan_capi, following some councelling from the gurus. I compiled it up, and when i try to load it in modules.conf, i get that wonderful message and Asterisk does not start: [chan_capi.so]Feb 17 09:21:40 WARNING[16384]: loader.c:239 ast_load_resource: /usr/lib/asterisk/modules/chan_capi.so: undefined symbol: ast_get_group Feb

hi all, I am trunking via iax2 2 asterisk serverses if both of them have static ip addresses, I can connect them using no password, password or auth rsa with a pair of keys. If one of them has dynamic ip address and need to register on the other server, I can connect them with no password, but I am not able to do that using keys. The question is: which is the right register syntax to use when

I am having a strange problem. On one * box I setup the fax recive, via spandsp -app_rxfax I have no problem here. On a second box I did the same. The resulting PDF appear "corrupt". If I transmit the same fax to both * box, the tiff files received are the same. A deeper analysis shows the only problem is the width and heigth of the document In the first PDF, I see

It seems to never fail - after 3 to 5 minutes SIP -> IAX calls drop audio on one side. I place a call out through voipjet, and call quality is flawless. However a few minutes later the person who I'm talking to can no longer hear me. I can still hear them. What should I look for to resolve this? Has anyone else had this problem? Using last night's CVS this problem still exists.

Does anyone have a document I can use as a guide on how to load a SIP image on a cisco 7960 phone? Ryan

If you define a sip peer, wheather or not you put an entry in the field OUTBOUND CID, if you dial an external extension (let's say an extension on another asterisk server, connected via IAX2 connection) the callerid received by the foreign asterisk is device <YOURNUMBER>: i.e device <567> If you take a look at etc/asterisk/sip_additional.conf, you can see under the SIP extension